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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
108 lines
3.2 KiB
C++
108 lines
3.2 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/rms_level.h"
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#include <math.h>
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#include <algorithm>
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#include <numeric>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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static constexpr float kMaxSquaredLevel = 32768 * 32768;
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// kMinLevel is the level corresponding to kMinLevelDb, that is 10^(-127/10).
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static constexpr float kMinLevel = 1.995262314968883e-13f;
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// Calculates the normalized RMS value from a mean square value. The input
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// should be the sum of squared samples divided by the number of samples. The
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// value will be normalized to full range before computing the RMS, wich is
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// returned as a negated dBfs. That is, 0 is full amplitude while 127 is very
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// faint.
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int ComputeRms(float mean_square) {
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if (mean_square <= kMinLevel * kMaxSquaredLevel) {
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// Very faint; simply return the minimum value.
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return RmsLevel::kMinLevelDb;
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}
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// Normalize by the max level.
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const float mean_square_norm = mean_square / kMaxSquaredLevel;
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RTC_DCHECK_GT(mean_square_norm, kMinLevel);
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// 20log_10(x^0.5) = 10log_10(x)
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const float rms = 10.f * log10(mean_square_norm);
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RTC_DCHECK_LE(rms, 0.f);
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RTC_DCHECK_GT(rms, -RmsLevel::kMinLevelDb);
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// Return the negated value.
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return static_cast<int>(-rms + 0.5f);
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}
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} // namespace
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RmsLevel::RmsLevel() {
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Reset();
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}
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RmsLevel::~RmsLevel() = default;
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void RmsLevel::Reset() {
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sum_square_ = 0.f;
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sample_count_ = 0;
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max_sum_square_ = 0.f;
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block_size_ = absl::nullopt;
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}
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void RmsLevel::Analyze(rtc::ArrayView<const int16_t> data) {
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if (data.empty()) {
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return;
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}
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CheckBlockSize(data.size());
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const float sum_square =
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std::accumulate(data.begin(), data.end(), 0.f,
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[](float a, int16_t b) { return a + b * b; });
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RTC_DCHECK_GE(sum_square, 0.f);
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sum_square_ += sum_square;
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sample_count_ += data.size();
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max_sum_square_ = std::max(max_sum_square_, sum_square);
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}
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void RmsLevel::AnalyzeMuted(size_t length) {
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CheckBlockSize(length);
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sample_count_ += length;
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}
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int RmsLevel::Average() {
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int rms = (sample_count_ == 0) ? RmsLevel::kMinLevelDb
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: ComputeRms(sum_square_ / sample_count_);
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Reset();
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return rms;
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}
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RmsLevel::Levels RmsLevel::AverageAndPeak() {
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// Note that block_size_ should by design always be non-empty when
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// sample_count_ != 0. Also, the * operator of absl::optional enforces this
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// with a DCHECK.
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Levels levels = (sample_count_ == 0)
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? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
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: Levels{ComputeRms(sum_square_ / sample_count_),
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ComputeRms(max_sum_square_ / *block_size_)};
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Reset();
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return levels;
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}
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void RmsLevel::CheckBlockSize(size_t block_size) {
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if (block_size_ != block_size) {
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Reset();
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block_size_ = block_size;
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}
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}
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} // namespace webrtc
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